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(Fixes AST-2007-026)
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added
via the manager. If a negative value is submitted for a member penalty, we set it to 0.
(closes issue #11411, reported and patched by Laureano)
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Reported by: chewbacca
Patch by: casper
(Closes issue #11401)
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The ast_set_callerid() function needed to lock the channel. Also, the handlers
for the CALLERID() dialplan function needed to lock the channel when reading
or writing callerid values directly on the channel structure.
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This makes ast_stopstream() thread-safe.
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(closes issue #11405)
Reported by: eliel
Patches:
load_realtime.patch uploaded by eliel (license 64)
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users.conf (reported internally, patched by me)
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every priority
executed in the dialplan if you have debug set to anything non-zero. This seems pointless
due to the fact that these channel variables are not referenced anywhere else in the code and
their names are esoteric enough that they would not be practical to reference in the dialplan. Plus
the fact that this behavior isn't documented anywhere means that the change is not likely to cause
any disruption. If anything, this may actually cause a slight performance increase if running with
debug on.
The motivating influence for this code change is the eventwhencalled option for queues. If set to
vars, all channel variables will be output to the manager. These unnecessary channel variables make
the output a lot more difficult to deal with.
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be stored.
Since greetings are not retrieved from IMAP anyway, it is pointless to attempt storing them there.
(closes issue #11359, reported by spditner, patched by me)
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handle locking the channel as needed
- update ast_explicit_goto() to lock the channel as needed
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eventwhencalled variable to 1.
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1) Due to some signed vs. unsigned silliness, setting 'eventwhencalled' to
'vars' or 'yes' did exactly the same thing. Thus the sign change of the
ast_true call.
2) The vars2manager function overwrote a \n for every channel variable it parsed, resulting
in bizarre output for the channel variables. This patch remedies this.
(related to issue #11385, however I'm not sure if this will actually be enough to close it)
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This set of changes fixes an issue that was reported to me on IRC yesterday.
The user, d1mas, was using chan_zap for incoming calls and was having DTMF
recognition issues in some situations. Specifically, he noticed that the
problem occurred when using DISA or WaitExten. He also noticed that when
using Read, the problem did not occur. His system also used DUNDi for
dialplan lookups.
So, he theorized that if the DUNDi lookups blocked for some period of time,
that audio from the zap channel could get lost. If the audio got lost, then
it wouldn't be run through the DTMF detector, and digits could get lost.
He was correct, and the following set of changes fixes the problem. However,
the changes go a little bit further than what was necessary to fix this exact
problem.
1) I updated pbx_extension_helper() to autoservice the associated channel to
handle cases where extension lookups may take a long time. This would
normally be a dialplan switch that does some lookup over the network, such
as the DUNDi or IAX2 switches.
This ensures that even while a DUNDi lookup is blocking, the channel will be
continuously serviced.
2) I made a change to the autoservice code. This is actually something that
has bothered me for a long time. When a channel is in autoservice, _all_
frames get thrown away. However, some frames really shouldn't be thrown
away. The most notable examples are signalling (CONTROL) frames, and DTMF.
So, this patch queues up important frames while a channel is in autoservice.
When autoservice is stopped on the channel, the queued up frames get stuck
back on the channel so that they can get processed instead of thrown away.
3) I made another change to the autoservice code to handle the case where
autoservice is started on channels recursively.
Previously, you could call ast_autoservice_start() multiple times on a
channel, and it would stop the first time ast_autoservice_stop() gets
called. Now, it will ensure that autoservice doesn't actually stop until
the final call to ast_autoservice_stop().
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ast_calloc().
(closes issue #11390, reported and patched by Laureano)
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parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL
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ignored due to lack of an argument
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only the first format specified will be attached to the message.
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Reported via the -users mailing list, fixed by me.
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another
codec that we don't know, Asterisk did not remove that codec from the list.
With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.
(closes issue #11376)
Reported by: lasse
Patches:
bug11376.txt uploaded by oej (license 306)
Tested by: lasse
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(closes issue #11304)
Reported by: pj
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cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
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playback of a number,
app_playback will continue to try to play the remaining files. With this change, no more files will
be played back upon hangup.
(closes issue #11345, reported and patched by IgorG)
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out to the CLI every time do_say in app_playback is called. Removing these
warnings
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has to be terminated and control returned to the thread before hanging things up. (issue #BE-252)
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(closes issue #11333)
Reported by: Laureano
Patches:
res_features_v2.c.patch uploaded by Laureano (license 265)
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little something I noticed while working on a completely unrelated issue.
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enabled.
(closes issue #11347)
Reported by: ys
Patches:
pbx.pbx_config.c.diff uploaded by ys (license 281)
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(closes issue #11357)
Reported by: reformed
Patches:
mixmonitor.patch uploaded by reformed (license 330)
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the option was supplied unless a non-zero-length argument was found for it
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was called. This prevents sending a duplicate e-mail.
(closes issue #11204, reported by spditner, patched by me)
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free the scheduler context.
(closes issue #11355)
Reported by: eliel
Patches:
main.channel.c.patch uploaded by eliel (license 64)
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(closes issue #11367)
Reported by: eliel
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a backslash. Unfortunately, this does not universally work on all databases,
since on databases which natively use the backslash as a delimiter, the
backslash itself needs to be delimited, but on other databases that have no
delimiter, backslashing the backslash causes an error.
So the only solution that I can come up with is to create an option in res_odbc
that explicitly specifies whether or not backslash is a native delimiter. If
it is, we use it natively; if not, we use the ESCAPE clause to make it one.
Reported by: elguero
Patch by: tilghman
(Closes issue #11364)
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Reported by: Laureano
Patch by: Laureano,tilghman
(Closes issue #11351)
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This change fixes the problem, with a multi-faceted approach. First, we
do our best to avoid these messages from being created in the first place,
and second, if that fails, we detect when the voicemail message is
zero-length and avoid exiting at that point.
Reported by: dtyoo
Patch by: gkloepfer,tilghman
(Closes issue #11083)
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invalid, due to the repetition of certain parameters in a single event.
This caused various issues for XML parsers, some of which refused to parse
at all, given the invalidity of the rendered XML. So this commit fixes
the XML output, ensuring that each entity parameter has a unique name, thus
ensuring valid XML.
Reported by: msetim
Patch by: tilghman
(Closes issue #10220)
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Reported by: apsaras
Patch by: tilghman
(Closes issue #11353)
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I was at it, I went ahead and tweaked it a little bit more.
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circumstances, this
message would always report that there were 0 members available, even though that may not be true.
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announcement to the masqueraded channel, and dial back to the original
channel on timeout.
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a string-based storage and comparison mechanism because it is easier to support on other platforms
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lock
the conlock as well as the hints lock, it must be locked in that respective order.
In order to prevent a potential deadlock, we need to lock the conlock prior to
locking the hints lock in ast_hint_state_changed (see the call stack example on
issue #11323 for how this can happen).
(closes issue #11323, reported by eelcob, suggestion for patch by eelcob, patch by me)
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